CN108370467B

CN108370467B - Control of sound effects in a room

Info

Publication number: CN108370467B
Application number: CN201680063550.5A
Authority: CN
Inventors: 尤哈·乌尔霍宁
Original assignee: Genelec Oy
Current assignee: Genelec Oy
Priority date: 2015-09-02
Filing date: 2016-08-31
Publication date: 2020-07-28
Anticipated expiration: 2036-08-31
Also published as: US20180268802A1; EP3345404B1; EP3345404A1; DK3345404T3; US10490180B2; EP3345404A4; FI20155634A; FI129335B; WO2017037341A1; ES2915843T3; CN108370467A

Abstract

According to an exemplary aspect of the invention, an apparatus is proposed, which comprises at least one processing core, and at least one memory including computer program code, the at least one memory and the computer program code being configured by the at least one processing core to enable the apparatus at least to: deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a loudspeaker for a first location of a first room; determining for the samples contained in the second digital information at least one of a delay value describing how much the time of the samples is to be moved and a gain factor describing how much the amplitude of the samples is to be adjusted such that the energy associated with the superposition of the system impulse response and the cancellation impulse response is reduced, and generating a cancellation signal using the determined at least one of the delay value and the gain factor and superposing the cancellation signal into the signal provided to the loudspeaker.

Description

Control of sound effects in a room

Technical Field

The invention relates to the field of loudspeakers and the control of at least part of at least one effect in a room.

Background

Equalization of the sound signal provided to the speaker may be used to control the frequency response. For example, when a speaker is to output sound in a first frequency range with a lower amplitude relative to other frequencies, the first frequency range is amplified in the input signal to compensate for the performance of the speaker and produce a more uniform frequency response.

Alternatively or in addition to making the frequency response more uniform, equalization may be used to modify the frequency response to conform to aesthetic preferences and/or characteristics of the environment in which the speaker is located. For example, certain types of music may be played out with lower frequencies emphasized if more information is included in the lower frequencies.

When listening to sound (e.g., music) output from at least one speaker in a room, the characteristics of the room may affect the listening experience. Such characteristics include the physical dimensions of the room, the materials used in the walls, roof and floor, and the objects (e.g., furniture) disposed in the room. In addition, the location of the speakers and listeners relative to the room and to each other may also affect the listening experience. The listening experience is significantly different at different locations in the room.

The use of loudspeakers in a room may cause resonance, also referred to as room effect, when the room is capable of reflecting sound, or effect in the space enclosed by walls and/or roof and floor. The effect will occur at a frequency corresponding to a multiple of a half wavelength, whereby the dimensions of the room define the frequency at which resonance may occur in the room. The smaller the room, the higher the frequency of the lowest effect. Thus, a larger room will have the lowest resonance that is so low that it is difficult for humans to hear. In addition, just as effects may occur in frequencies related to room size, effects may also occur in higher frequencies. The density of the room effect (defined as the number of room effects per unit frequency) increases with increasing frequency.

The room effect may store energy. The energy of the sound output through the speaker is stored in the vibrations sustained by this effect. If the speaker is turned off, the room effect will diminish at a rate determined by the room's absorptivity (among others). Since room effects may affect the listening experience, it is useful to control the energy stored in the effects. The control method includes the design of the room geometry, covering the walls with a material that is absorptive in the frequencies where the effect resonances exist, and equalizing the speaker output so that the output in the effect frequencies is reduced so that the sound level in the room effect does not increase more than in the frequencies near the effect, thereby equalizing the volume changes affected by the effect.

Document WO 99/66492 discloses a sound reproduction apparatus for reducing the level of sound reflections in a room. Document EP 1322037 discloses a method for designing a modal balancer for low frequency sound reproduction. Document EP 1516511 discloses a method for designing a modal balancer for a low frequency audible range, in particular for a close-fitting mode.

Disclosure of Invention

According to a first aspect of the present invention, there is provided an apparatus comprising at least one processing core and at least one memory including computer program code, the at least one memory and the computer program code configured by the at least one processing core to enable the apparatus at least to: deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a loudspeaker for a first location of a first room; determining for the samples contained in the second digital information at least one of a delay value describing how much the time of the samples is to be moved and a gain factor describing how much the amplitude of the samples is to be adjusted such that the energy associated with the superposition of the system impulse response and the cancellation impulse response is reduced, and generating a cancellation signal using the determined at least one of the delay value and the gain factor and superposing the cancellation signal into the signal provided to the loudspeaker.

Many embodiments of the first aspect may include at least one feature from the following list:

the signal supplied to the loudspeaker comprises a payload signal

The at least one memory and the computer program code configured, with the at least one processing core, to further cause the apparatus to receive first digital information from a sensor

The at least one memory and the computer program code are configured, with the at least one processing core, to further enable the apparatus to derive second digital information at least in part by multiplying respective samples contained in the first digital information by coefficient values

The coefficient value comprises a negative value

The at least one memory and the computer program code are configured, with the at least one processing core, to further cause the apparatus to low pass filter the first digital information prior to deriving the second digital information

The at least one memory and the computer program code are configured, with the at least one processing core, to further cause the apparatus to determine at least one of a second delay value and a second gain factor from a second system impulse response and a second cancellation impulse response, the second system impulse response being generated by a second location of the speaker for the first room, and to cause the apparatus to superimpose the cancellation signal or a second cancellation signal according to at least one of the second delay value and the second gain factor into a signal provided to the speaker according to an operator selection

The at least one memory and the computer program code configured to, with the at least one processing core, enable the apparatus to determine at least one of the delay value and the gain factor by using an optimization algorithm

The optimization algorithm comprises at least one of a least squares method, a simulated annealing algorithm and a direct search algorithm

The system impulse response and the cancellation impulse response together form an equalized system impulse response

The system impulse response and the cancellation impulse response together form a reproduction system, wherein for at least one resonance, the resonance in the room is removable by canceling the acoustic reflections by means of at least one wall.

According to a second aspect of the invention, there is provided an apparatus comprising: a receiver configured to receive an input signal; a delay buffer configured to delay a first copy of an input signal by a configurable delay length to produce a cancellation signal; and an addition circuit configured to add a sample contained in the cancellation signal to a sample contained in the second copy of the input signal.

Many embodiments of the second aspect may include at least one feature from the following list:

in generating the cancellation signal, the apparatus is further configured to multiply samples from the delay buffer by a coefficient value, wherein the coefficient value effectively inverts the phase of the signal

The coefficient value comprises a negative value

The apparatus is configured to enable sample inversion and amplitude scaling from the delay buffer

The apparatus also includes a low pass filter configured to receive and low pass filter a first copy of the input signal and provide the low pass filtered first copy of the input signal to the delay buffer

The apparatus further comprises a high pass filter arranged to receive and high pass filter a second copy of the input signal and to provide a high pass filtered second copy of the input signal to the summing circuit.

According to a third aspect of the invention, a method is proposed, the method comprising: deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a loudspeaker for a first location of a first room; determining for the samples contained in the second digital information at least one of a delay value describing how much the time of the samples is to be moved and a gain factor describing how much the amplitude of the samples is to be adjusted such that the energy associated with the superposition of the system impulse response and the cancellation impulse response is reduced, and generating a cancellation signal using the determined at least one of the delay value and the gain factor and superposing the cancellation signal into the signal provided to the loudspeaker.

Many embodiments of the third aspect may comprise at least one feature corresponding to a feature in the list listed in association with the first aspect.

According to a fourth aspect of the invention, a method is presented, the method comprising: receiving an input signal; delaying the first copy of the input signal in a delay buffer by a configurable delay length to produce a cancellation signal; and adding samples contained in the cancellation signal to samples contained in a second copy of the input signal.

Many embodiments of the fourth aspect may comprise at least one feature corresponding to a feature in the list listed in association with the second aspect.

According to a fifth aspect of the invention, there is provided an apparatus comprising: means for deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a speaker for a first location of a first room; means for determining, for a sample contained in the second digital information, at least one of a delay value and a gain factor, the delay value describing how much the time of the sample is to be moved, the gain factor describing how much the amplitude of the sample is to be adjusted such that the energy associated with the superposition of the system impulse response and the cancellation impulse response is reduced, and means for generating a cancellation signal using the determined at least one of the delay value and the gain factor and superposing the cancellation signal into the signal provided to the loudspeaker.

According to a sixth aspect of the present invention, there is provided an apparatus comprising means for receiving an input signal; a mechanism for delaying a first copy of an input signal by a configurable delay length to produce a cancellation signal; and means for adding samples contained in the cancellation signal to samples contained in the second copy of the input signal.

According to a seventh aspect of the invention, there is provided a non-transitory computer readable medium having stored therein a set of computer readable instructions which, when executed by at least one processor, are capable of causing an apparatus at least to: deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a loudspeaker for a first location of a first room; determining for the samples contained in the second digital information at least one of a delay value describing how much the time of the samples is to be moved and a gain factor describing how much the amplitude of the samples is to be adjusted such that the energy associated with the superposition of the system impulse response and the cancellation impulse response is reduced, and generating a cancellation signal using the determined at least one of the delay value and the gain factor and superposing the cancellation signal into the signal provided to the loudspeaker.

According to an eighth aspect of the invention, there is provided a non-transitory computer readable medium having stored therein a set of computer readable instructions which, when executed by at least one processor, can cause an apparatus at least to: receiving an input signal; delaying the first copy of the input signal by a configurable delay length to produce a cancellation signal; and adding samples contained in the cancellation signal to samples contained in a second copy of the input signal.

According to a ninth aspect of the present invention, there is provided a computer program configured to be capable of performing the method according to any one of the third or fourth aspects.

Industrial applications

At least certain embodiments of the present invention may be used for improved acoustic resonance cancellation within an enclosed space where resonance cancellation can be successful and/or acoustic resonance cancellation in a larger area.

Drawings

FIG. 1 illustrates an exemplary system capable of supporting at least some embodiments of the invention;

FIG. 2 shows the attenuation of the volume of room effect frequencies;

FIG. 3 illustrates an exemplary apparatus capable of supporting at least certain embodiments of the invention;

FIG. 4 illustrates the use of delay values and gain factors in accordance with at least some embodiments of the present invention;

FIG. 5 illustrates a second use of delay values and gain coefficients in accordance with at least some embodiments of the present invention;

FIG. 6 illustrates the use of at least two delay values and gain factors in accordance with at least some embodiments of the present invention;

FIG. 7 shows an exemplary framework in accordance with at least some embodiments of the invention;

FIG. 8 shows the attenuation of volume in the room effect when using effect cancellation;

FIG. 9 shows the attenuation of the volume in the room effect with and without room effect cancellation;

FIG. 10 shows a first flowchart of a method in accordance with at least some embodiments of the invention;

fig. 11 shows a second flowchart of a method in accordance with at least some embodiments of the invention.

Detailed Description

By introducing a delayed and attenuated copy of the input signal into the input signal, the room effect can be controlled by having the loudspeaker output the same frequency but opposite phase sound into the room effect after a certain time before providing the input signal to the loudspeaker element. This may help to actively absorb energy from the room effect to at least partially cancel sound. By using the same loudspeaker to produce sound and canceling sound of opposite phase, the effect area of the effect cancellation is enhanced compared to a solution in which a separate loudspeaker is used to output canceling sound for controlling room effects. Since the canceling sound radiated according to the present invention is radiated with exactly the same physical location and sound structure as the original audio, the sound effects (e.g., delay, reflection, and reverberation) experienced by the canceling sound are expressed in the same manner as the original sound. This can enhance the effect of the effect control according to the present invention.

FIG. 1 illustrates an exemplary system capable of supporting at least some embodiments of the invention. Fig. 1 shows a room 100, which may be constructed of brick, cement, cardboard, or other material suitable for constructing a room. Such materials typically exhibit acoustic reflection capabilities. Within the room 100, a speaker 110 is provided, which speaker 110 may comprise, for example, a subwoofer, a woofer, a midrange speaker, or a full range speaker, or combinations thereof. The speaker 110 may be based on, for example, electrodynamic, magnetic, magnetostrictive, electrostatic, or other techniques. The speaker 110 may be configured to produce sound based on an input signal that encodes the sound in a suitable format.

The speaker 110 may emit sound in all directions or only at least partially directionally. In some speakers, the directionality of the emitted sound depends at least in part on the frequency. For example, low frequencies may be emitted more omnidirectionally than higher frequencies. Arrow 112 represents sound emitted from speaker 110, and arrow 114 represents a reflection of at least a portion of the sound represented by arrow 112. The sound reflections may cause room effects, such that direction 120 defines some room effect frequencies. To the extent that the speaker 110 emits sound in these frequencies, the energy associated with the sound may have a tendency to accumulate in room effects.

The location 130 represents a location in the room 100 where the listener can hear the sound output from the speaker 110. The setting of the location 130 in the room 100 in fig. 1 is merely illustrative and does not limit the locations at which the location 130 may be set in the room 100.

Fig. 2 shows the attenuation of the volume of the room effect frequency. In the displayed coordinate system, time grows linearly from top to bottom. The frequency grows logarithmically from left to right. The volume is gray scale encoded in the image.

In the shown figure, the room effect can be seen at a frequency of 45 Hz. The 60dB volume attenuation stored in the effect takes about 500 milliseconds. Such effects may negatively impact the listening experience if the listener is located in a room where the effect can be heard.

To control the 45Hz effect, a speaker (e.g., speaker 110 in fig. 1) may be configured to emit less energy at a frequency of 45Hz, for example, by equalization. This reduces the volume of the proximity effect compared to the case where amplitude equalization is not used. However, this conventional amplitude equalization does not reduce the rate of attenuation of the effective volume. The attenuation rate is a rate at which sound pressure decreases after sound input by the speaker stops. Furthermore, it is difficult to achieve a mathematically complete cancellation that can completely suppress the effects of room effect resonances and thus this approach is deficient.

Active cancellation of sound includes, for example, emitting sound from a speaker element that adds up to the sound in the opposite phase to reduce the volume. Since the original sound as well as the canceling sound are propagated to all directions in the three-dimensional space, the sound emitted from the speaker element for canceling the original sound may cause only partial cancellation in the room, whereby cancellation does not occur equally in the space of the entire room. This is because the phases of the original sound and the canceling sound cannot be in completely opposite phases throughout the room. For example, by providing a microphone (which may be considered a second speaker) proximate to the speaker that radiates canceling sound, the canceling may be arranged to occur locally in the room. A negative feedback loop can be effectively provided from the signal picked up by the microphone to the signal radiated by the loudspeaker to reduce the volume. A band pass filter may be used to limit this setting to a certain band pass frequency. This arrangement creates a dead space near the second speaker to effectively reduce acoustic resonances and change the apparent amount of absorption by the walls of the room near the second speaker. This arrangement may reduce certain room effects, typically only in proximity to the second speaker, and in this way is limited compared to the proposed technique.

To improve the listening experience at the location 130, one or more room effects may be attempted to be eliminated, so that the effect of the elimination may be produced at least at the location 130. If the room effect does not cancel to the same extent elsewhere in the room 100, this may have little effect on the listener if the listener is listening at the location 130. To help improve the listening experience, the elimination of room effects need not be complete, as partial elimination may already provide a significant improvement in the listening experience.

For cancellation, a separate speaker element may be used to emit cancellation sound configured such that the effect of the sound emitted from the separate speaker element at least partially cancels the effect of the room effect at location 130. Alternatively, the same speaker element may be used to produce both the original sound for listening and the canceling sound.

To enable such cancellation, the speaker 110 (or a control element of the speaker 110) may provide a system impulse response of the speaker 110 at the location 130, i.e., an impulse response that the speaker 110 brings to the location 130 by emitting sound from the speaker 110 at the location in the room 100. For example, a microphone may be provided at location 130 to record the sound emitted by the speaker, and the recorded information (e.g., in digitized format) may be provided to the speaker 100 or a control element to facilitate determination of the system impulse response.

The system impulse response may be low pass filtered before being provided to the speaker or controller in anticipation of the room effect to be controlled occurring in low frequencies. In the loudspeaker or control element, the system impulse response may be inverted.

When the system impulse response is reversed, the delay and/or amplitude magnitudes can be varied to determine a delay and/or gain factor that minimizes the energy of the superposition of the original system impulse response and the delay and/or gain adjusted impulse response at location 130. Such a scaled and/or delayed impulse response is referred to as a cancellation impulse response. The superposition energy may be determined as the sum of the squares of the original system impulse response and the cancellation impulse response, and a minimization algorithm may be used to minimize this quadratic expression. The energy is related to the boost stage and the audio signal level. Examples of suitable minimization algorithms include the simplex algorithm, the newton method, and the Nelder-Mead method. More generally, an optimization algorithm, such as a simulated annealing algorithm or a direct search algorithm, may be used.

In certain embodiments, the use of an optimization algorithm (e.g., a minimization algorithm) may not be necessary if the speaker or control element can determine a corrected delay value and gain factor for cancellation at location 130, for example, by the delay value and amplitude of the first peak in the system impulse response vector. For example, the delay value may be considered as the delay value associated with the first early reflection peak in the system impulse response, and the gain factor may be set to-3 dB, for example.

In other embodiments, the speaker 110 or the control element may be configured to experimentally determine the delay value and/or the gain factor. In these embodiments, the speaker and control element may reverse the recording and then emit the test sound. The test sound may include a copy of the original sound as well as a delayed and/or inverted original sound. In this manner, the speaker or control element may experimentally determine which delay value and gain factor may produce the lowest energy superimposed impulse response at location 130.

Regardless of the manner in which the delay value and/or gain factor is determined, the determined delay value and/or gain factor may be used in the speaker when the latter sound is played. Fig. 4 shows an embodiment using delay values and/or gain factors. In fact, the use of a cancellation impulse response and a system impulse response as described above may be considered to produce an equalized system impulse response. The determination of the delay value and/or the gain factor may occur in the speaker, a speaker controller, or a control device (e.g., a processor) configured to control the speaker or speaker controller when implanted therein.

In some embodiments, more than one location in the room 100 may be used to determine the delay value and/or gain factor. For example, if the room has a second listening position, the delay value and/or gain factor may also be confirmed for the second position. The user may then select which delay value and/or gain factor to use to optimize performance at the desired listening position. Since the cancellation of the room effect may be location specific, the room effect at location 130 may not be effectively cancelled when using the second location. Also, room effects in the second location may not be effectively eliminated when using location 130.

Determining the delay value and/or gain factor as described above may enable room effect cancellation or at least partial reduction in a room having a complex shape or comprising objects that make it difficult to analytically calculate sound propagation and room characteristics. It is useful to use the same speaker elements to play the original sound and cancel the sound for listening at location 130, since the sound interacts with the room in the same way as it is emitted from the same device in the same direction. An additional advantage of at least some embodiments of the invention is that a separate cancellation speaker is not required. The present invention is advantageous over using a separate cancellation speaker because the original sound and the cancellation sound can be reproduced identically and interact with the room in the same way. Thus, the room 100 space where sufficient effect cancellation can occur can be large when using the present invention.

In general, certain embodiments of the present invention do not require scaling and/or inversion, but merely apply a delay to achieve a cancellation impulse response. Thus, any additional phases described above are not mandatory features of the invention. For example, in a narrow band implementation, proper selection of the delay value may result in cancellation of the impulse response. In this case, an optimization method or other methods may be used to determine the delay value. The optimization may be easier and faster when only one parameter is optimized. On the other hand, in some embodiments, more than one delay value and/or gain factor may be selected to obtain a cancellation impulse response.

Fig. 3 illustrates an exemplary device capable of supporting at least some embodiments of the present invention. Shown is an apparatus 300, which may include, for example, a speaker or a controller for a speaker. Included in the apparatus 300 is a processor 310, which may include, for example, a single-core processor including one processing core or a multi-core processor including more than one processing core. Processor 310 may comprise, for example, a highpass cellcell 800 processor. The processor 310 may include more than one processor. The processing core may include, for example, a Cortex-A8 processing core manufactured by Intel Corporation, or a Brisbane processing core manufactured by Advanced micro devices Corporation. The processor 310 may include at least one application specific integrated circuit, i.e., ASIC. The processor 310 may include at least one field programmable gate array, i.e., FPGA. The processor 310 may be a means for performing method steps in the apparatus 300. The processor 310 may be a means for performing method steps in the apparatus 300. Processor 310 may be configured to perform actions at least in part by computer instructions.

The apparatus 300 may include a memory 320. Memory 320 may include random access memory and/or persistent memory. The memory 320 may include at least one RAM chip. The memory 320 may include, for example, magnetic memory, optical memory, and/or holographic memory. The memory 320 is at least partially accessible to the processor 310. The memory 320 may be a device for storing information. The memory 320 may include computer instructions that the processor is configured to execute. Where computer instructions configured to cause processor 310 to perform certain actions are stored in memory 320 and device 300 is configured generally to be operable under the direction of processor 310 using the computer instructions from memory 320, processor 310 and/or at least the processing cores thereof may be considered to be configured to perform certain actions as described above.

The apparatus 300 may include a transmitter 330. The apparatus 300 may include a receiver 340. Transmitter 330 and receiver 340 may be configured to transmit and receive information, respectively, according to at least one standard. The transmitter 330 may include more than one transmitter. Receiver 340 may include more than one receiver. Transmitter 330 and/or receiver 340 may be configured to operate according to an ethernet and/or communication bus internal to the speaker. Transmitter 330 and/or receiver 340 may be configured to communicate using digital or analog techniques, for example.

The apparatus 300 may include a Near Field Communication (NFC) transceiver 350. NFC transceiver 350 may support at least one NFC technology, such as NFC, bluetooth, Wibree, or similar technologies.

The apparatus 300 may include a User Interface (UI) 360. UI 360 may include at least one of a display, a keyboard, a touch screen, a vibrator configured to send a signal to a user by causing apparatus 300 to vibrate, a speaker, and a microphone. The user can operate the device 300 through the UI 360, for example, to select inputs, sound output formats, and/or to select listening positions to be used in the room.

The processor 310 may be equipped with a transmitter configured to output information from the processor 310 to other devices included in the apparatus 300 via electrical leads internal to the apparatus 300. Such a transmitter may comprise a serial bus transmitter, for example, configured to output information to memory 320 via at least one electrical lead for storage therein. As an alternative to a serial bus, the transmitter may comprise a parallel bus transmitter. Likewise, the processor 310 may include a receiver configured to receive information in the processor 310 from other devices included in the apparatus 300 via electrical leads internal to the apparatus 300. Such a receiver may comprise a serial bus receiver, for example, configured to receive information from receiver 340 over at least one electrical lead for processing in processor 310. As an alternative to a serial bus, the receiver may comprise a parallel bus receiver.

Processor 310, memory 320, transmitter 330, receiver 340, NFC transceiver 350, and/or UI 360 may be connected to each other in many different ways by electrical leads internal to device 300. For example, the aforementioned devices may each be separately connected to a main bus internal to the apparatus 300 to allow information to be exchanged between the devices. However, the skilled person will understand that this is not the only example and that many ways of interconnecting at least two of the aforementioned devices may be chosen according to the embodiments without departing from the scope of the invention.

Fig. 4 shows the use of delay values and gain factors, for example in sub-woofer elements. In stage 410, an input signal is received, wherein the input signal may comprise, for example, a digital input signal. The received input signal is replicated into two copies, the first copy being provided to stage 420, where the first copy is delayed by the determined delay value. The delayed first copy is provided from stage 420 to stage 430, where the amplitude of the first copy is modified by a coefficient value (e.g., one in the interval [ -1,0 ]), to result in delayed first copy inversion and amplitude scaling.

In stage 440, the delayed, inverted and scaled copy from stage 430 is superimposed with a second copy of the input signal to produce a modified signal. In stage 450, the modification signal is provided to a speaker. In some embodiments, the input signal is analog in nature, rather than digital. If the loudspeaker element is a sub-woofer loudspeaker element, low pass filtering may not be required, since the sub-woofer loudspeaker is intended to operate over a narrow frequency band and any phase shift caused by low pass filtering is avoided when generating the delayed first copy of the input signal.

Fig. 5 shows a second use of the delay values and gain factors, for example in the main loudspeaker. Fig. 5 may occur, for example, in the digital domain.

Stages

410, 420, 430, 440 and 450 here correspond substantially to the identically numbered stages in fig. 4. Fig. 5 additionally includes a stage 510 in which the first copy of the input signal is low pass filtered before being delayed. A copy of the low pass filtered input signal may be provided from stage 510 to stage 440 where the first copy, the second copy, and the copy from stage 510 are added together. The superimposing may be achieved, for example, by superimposing digital samples. Alternatively, the low pass filtering may be performed after a delay (e.g., between

stages

420 and 430, or between stages 430 and 440). Fig. 5 also includes high pass filtering the second copy of the input signal prior to the mixing stage 440. In practice, the first copy corresponds to a low frequency of the input signal and the second copy corresponds to a high frequency of the input signal, wherein the delay and amplitude scaling of

stages

420 and 430 are implemented only for low frequencies. This is because the room effect is expected to occur mainly at low frequencies. This can result in the cancellation impulse response containing a limited frequency range (typically low frequencies) within which room effects need to be controlled.

Fig. 6 shows the use of at least two delay values and gain factors. In fig. 6, the input is provided to a first band pass filtering step 620 and a second band pass filtering step 650. One copy of the input may be provided directly to the superimposing step 680. In a first band pass filtering step 620, the input is band pass filtered and the result of the band pass filtering is provided to a delay step 630. In a delay step 630, a first delay value is applied to the result obtained by the first band-pass filtering step. The delayed samples are provided from the delay step 630 to a first inversion and scaling step 640, where the signal is scaled and/or inverted by a first gain factor to obtain a first cancellation signal. The first cancellation signal is provided from the first inverting and scaling step 640 to the superimposing step 680.

Likewise, a copy of the input is provided to a second bandpass filtering step 650. In a second bandpass filtering step 659, the input is bandpass filtered and the result of the bandpass filtering is provided to a delay stage 660. The band pass of the second band pass filtering step 650 may be different from the band pass of the first band pass filtering step 620. In a delay step 660, a second delay value is applied to the result obtained by the second band-pass filtering step 650. From the delay step 660 the delayed samples are provided to a second inversion and scaling step 670, where the signal is scaled and/or inverted by a second gain factor to obtain a second cancellation signal. The second cancellation signal is provided from the second inversion and scaling step 670 to the superposition stage 680.

Although the architecture shown contains two bandpass, delay and inversion and scaling stages, the architecture in fig. 6 may also include variations with different numbers, for example, there may be three, four or five bandpass, delay and inversion and scaling steps. In general, the arrangement of the various bandpass, delay, and inversion and scaling steps may be configured to provide one cancellation signal to the superimposing step 680. The bandpass, delay, and inversion and scaling arrangements beyond the second are shown schematically in figure 6 as arrangement 6100. Arrangement 6100 receives a copy of the input signal, derives a cancellation signal as shown above, and provides the cancellation signal to the superimposing step 680.

In a superimposing step 680, the copy of the signal received directly from the input 610 and the respective cancellation signals received from the inverting and scaling steps are superimposed to produce an output signal 690.

In the architecture in fig. 6, the following advantages are obtained: the cancellation signal is set to react to frequencies associated with room effects, while other frequencies are unaffected. In at least some embodiments, the band-pass of the band-pass filtering step is configurable. The room effect (and the band pass of the band pass filtering step) may be based on relatively low frequencies. In some embodiments, the number of arrangements of bandpass, delay, and inversion and scaling steps used may be configurable. For example, the number of these arrangements may be selected to match the number of room effects detected. For example, the apparatus may comprise a plurality of such arrangements, which are selectively put into use, or the apparatus may be processed by software to allow such arrangements to be dynamically generated.

Fig. 7 shows an exemplary architecture in accordance with at least some embodiments of the invention. In fig. 7, the input signal is provided to the high pass filtering step 720 in one copy and to the low pass filtering step 740 in one copy. In a high pass filtering step 720 the signal is high pass filtered and the result of the high pass filtering is provided to a superimposing step 790 by a delay step 725.

In a low pass filtering step 740 the signal is low pass filtered and the result of the low pass filtering is provided to a superimposing step 790 by a delay step 745. Another copy of the low pass filtered signal is provided to decimation step 750. In the decimation step 750, the number of samples per time unit is reduced, for example by a factor of 2 or 4. After the decimation step, the signal is delayed according to the delay value and provided to the inversion and scaling step 770 in a delay step 760. In an inversion and scaling step 770, the signal is scaled and/or inverted by a gain factor to obtain a cancellation signal. In general, scaling may include multiplying by a coefficient value. The cancellation signal is output from the inversion and scaling step 770 to the superposition step 790 through the interpolation step 780 where the sample frequency of the signal is restored, i.e. the decimation step 750 is inverted. The interpolation may be performed using an interpolation filter, for example. The interpolation step 780 may include a low pass filter function configured to remove images resulting from the interpolation. In a superimposing step 790, the output signal 7100 is obtained by superimposing the high pass filtered signal, the low pass filtered signal and the cancellation signal. The delays in delay steps 760, 725 and 745 need not be the same. In particular, in at least some embodiments, no two of the three delay steps will be the same. The delays of the delay steps 725 and 745 may be implemented to compensate for the delays introduced by the low pass filtering step 740, the delay step 760, the inversion and scaling step 770, and the decimation step 750 and the interpolation step 780 so that the signals arrive synchronously at the superposition step 790 to facilitate their superposition.

An advantage of deriving the cancellation signal in decimated form is that computational complexity can be reduced. In the decimated form, a sufficiently accurate cancellation signal may be obtained because it may have a relatively low frequency of room effects compared to many cases.

Fig. 8 shows the attenuation of room effects with effect cancellation (e.g. exemplary cancellation according to fig. 4, 5, 6 or 7). Fig. 8 shows the same room as in fig. 2, however one can immediately see that by eliminating, the level of the effect of 45Hz can be significantly reduced in times below 100 milliseconds. This has the following effect: the listening experience, for example at location 130, is significantly improved such that the attenuation is substantially inaudible.

Fig. 9 shows the attenuation of the room effect with and without cancellation. In

graphs

910 and 920, each of the plurality of drop lines corresponds to a signal measurement. Graph 910 shows the attenuation of the effect of 45Hz without cancellation. On the vertical axis is sound pressure level, expressed in decibels, and on the horizontal axis is time, in seconds. The attenuation of the effect is linear on the decibel scale, meaning that the volume decays exponentially on the linear scale, losing a constant percentage of energy in each reflection.

Fig. 920 shows the attenuation using the same effect according to an embodiment of the invention. The axis of the diagram 920 is similar to the axis of the diagram 910. As shown, the effect decays by more than 15dB in about 70-80 milliseconds, whereas without cancellation, the 15dB decay takes a considerable time, about 200 milliseconds, in graph 910.

Fig. 10 is a first flowchart of a method in accordance with at least some embodiments of the present invention. The steps of the method shown may be performed in a loudspeaker or a control element included in the loudspeaker to control its operation. Step 1010 includes deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a speaker for a first location in a first room. Step 1020 includes determining a delay value that describes how much the sample should be moved in time and/or a gain factor that describes how much the amplitude of the sample should be adjusted to minimize, or at least reduce, the energy associated with the superposition of the system impulse response and the cancellation impulse response. Finally, step 1030 includes generating a cancellation signal using the determined at least one of the delay value and the gain factor, and superimposing the cancellation signal into the signal provided to the speaker.

Fig. 11 shows a second flowchart of a method in accordance with at least some embodiments of the invention. The steps of the method shown may be carried out in a loudspeaker or a control element contained within a loudspeaker for controlling the operation thereof. Step 1110 includes receiving an input signal. Step 1120 includes delaying the first copy of the input signal in a delay buffer by a configurable delay length to produce a cancellation signal. The value of the delay length may be set, for example, to the determined delay value. Optional step 1130 includes multiplying the samples from the delay buffer by the coefficient value to produce a cancellation signal. In general, multiplying a sample by a coefficient value may include any processing sequence that has the overall effect of multiplying a sample by a coefficient value. Finally, step 1140 includes adding samples contained in the cancellation signal to samples contained in the second copy of the input signal. The method may also include providing a second copy of the input signal to the speaker after step 1140 to provide a version of the input signal to the speaker that has been modified by the superposition cancellation signal.

In general, the invention provides an apparatus comprising at least one processing core, at least one memory including computer program code, the at least one memory and the computer code configured, with the at least one processing core, to cause the apparatus at least to: deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a loudspeaker for a first location of a first room; determining, for the samples contained in the second digital information, at least one of a delay value describing how much the samples are time shifted and a gain factor describing how much the amplitude of the samples is to be adjusted to minimize, or at least reduce, the energy associated with the superposition of the system impulse response and the cancellation impulse response; and generating a cancellation signal using the determined delay value and amplitude and superimposing the cancellation signal into the signal provided to the loudspeaker. Deriving the gain factor may include selecting the gain factor, for example the gain factor may be selected to be-3 dB with respect to the signal provided to the loudspeaker. Deriving the second digital information may comprise, at least in part, deriving the first digital information by copying it. Generating the cancellation signal using the determined delay value and gain factor may include generating a copy of the signal, delaying the copy relative to the signal by the delay value, and adjusting an amplitude of the copy based on the amplitude.

It is to be understood that the disclosed embodiments of this invention are not limited to the particular structures, method steps, or materials disclosed herein, but extend to equivalents thereof as would be recognized by those ordinarily skilled in the relevant arts. It is also to be understood that the terminology used herein is for the purpose of describing particular embodiments only and is not intended to be limiting.

Reference herein to one embodiment means that a particular feature, structure, or characteristic described in connection with the embodiment is included in at least one embodiment of the invention. Thus, the appearances of the phrase "in one embodiment" appearing in various places throughout the specification are not necessarily all referring to the same embodiment.

As used herein, a plurality of items, structures, components and/or materials may be listed in a common list for convenience. However, these lists should be construed as though each element of the list is individually identified as a separate and unique element. Thus, no single element from such lists should be construed as a de facto equivalent of any other element from the same list solely based on their presentation in a common group without indications to the contrary of such list. Additionally, various embodiments and examples of the present invention may be considered herein as alternatives to its various components. It should be understood that these embodiments, examples and alternatives are not to be considered as actual equivalents of one another, but rather are to be considered as independent and autonomous manifestations of the invention.

Furthermore, the described features, structures, or characteristics may be combined in any suitable manner in one or more embodiments. In this description, numerous specific details are set forth, such as examples of lengths, widths, shapes, etc., in order to provide a thorough understanding of embodiments of the present invention. One skilled in the relevant art will recognize, however, that the invention may be practiced without one or more of the specific details, or with other methods, components, materials, and so forth. In other instances, well-known structures, materials, or operations are not shown or described in detail to avoid obscuring aspects of the invention.

While the foregoing embodiments illustrate the principles of the invention in one or more particular applications, it will be appreciated by those skilled in the art that numerous modifications in form, usage and details of implementation can be made without the exercise of inventive faculty, and without departing from the principles and concepts of the invention. Accordingly, it is not intended that the invention be limited, except as by the appended claims.

Claims

1. An apparatus (300) for controlling loudspeakers, comprising at least one processing core (310) and at least one memory (320) containing computer program code, the at least one memory (320) and the computer program code being configured by the at least one processing core (310) to enable the apparatus (300) at least to:

deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a loudspeaker for a first location (130) of a first room;

determining, for a sample contained in the second digital information, a delay value describing how much the time of the sample is to be moved and an attenuation coefficient describing how much the amplitude of the sample is to be reduced such that the energy associated with the superposition of the system impulse response and the cancellation impulse response is reduced, and

the determined delay value and attenuation coefficient are applied to a first copy of the input signal provided to the loudspeaker to produce a cancellation signal, and the cancellation signal is superimposed on the input signal provided to the loudspeaker.

2. The apparatus (300) of claim 1, wherein the signal provided to the speaker comprises a payload signal.

3. The apparatus (300) of claim 1 or 2, wherein the at least one memory (320) and the computer program code are configured, with the at least one processing core (310), to further cause the apparatus (300) to receive the first digital information from the sensor.

4. The apparatus (300) of claim 1, wherein the at least one memory (320) and the computer program code are configured, with the at least one processing core (310), to further enable the apparatus (300) to derive the second digital information at least in part by multiplying respective samples contained in the first digital information by coefficient values.

5. The apparatus (300) of claim 4, wherein the coefficient value comprises a negative value.

6. The apparatus (300) of claim 1, wherein the at least one memory (320) and the computer program code are configured, with the at least one processing core (310), to further cause the apparatus (300) to low pass filter the first digital information prior to deriving the second digital information.

7. The apparatus (300) of claim 1, wherein the at least one memory (320) and the computer program code are configured, with the at least one processing core (310), to enable the apparatus (300) to determine the delay value and the attenuation coefficient by using an optimization algorithm.

8. The apparatus (300) of claim 7, wherein the optimization algorithm comprises at least one of a least squares method, a simulated annealing algorithm, and a direct search algorithm.

9. The apparatus (300) of claim 1, wherein said system impulse response and said cancellation impulse response together form an equalized system impulse response.

10. An apparatus for controlling a speaker, comprising:

a receiver configured to receive an input signal;

it is characterized by also comprising:

a delay buffer configured to delay the first copy of the input signal by a configurable delay length to produce a cancellation signal, an

An adder circuit configured to superimpose samples contained in a cancellation signal onto samples contained in a second copy of an input signal, wherein in generating the cancellation signal, the apparatus is further configured to multiply samples from the delay buffer by a coefficient value, wherein the coefficient value effectively inverts a phase of the first copy of the input signal.

11. The apparatus of claim 10, further comprising a low pass filter configured to receive and low pass filter the first copy of the input signal and provide the low pass filtered first copy of the input signal to the delay buffer.

12. The apparatus of claim 10, further comprising a high pass filter configured to receive and high pass filter the second copy of the input signal and provide the high pass filtered second copy of the input signal to the summing circuit.

13. A method for controlling a speaker, comprising:

deriving second digital information describing a cancellation impulse response from first digital information describing a system impulse response generated by a speaker for a first location of a first room (1010);

determining for the samples contained in the second digital information a delay value describing how much the time of the samples is to be moved and a decay coefficient describing how much the amplitude of the samples is to be reduced such that the energy associated with the superposition of the system impulse response and the cancellation impulse response is reduced (1020), and

the determined delay value and attenuation coefficient are applied to a first copy of the input signal provided to the loudspeaker to produce a cancellation signal, and the cancellation signal is superimposed on the input signal provided to the loudspeaker (1030).

14. The method of claim 13, wherein the signal provided to the speaker comprises a payload signal.

15. The method of claim 13, further comprising receiving the first digital information from a sensor.

16. The method of claim 14 or 15, further comprising deriving the second digital information at least in part by multiplying respective samples contained in the first digital information by coefficient values, wherein the coefficient values effectively invert the phase of the first copy of the input signal.

17. A method for controlling a speaker, comprising:

receiving an input signal (1110);

it is characterized by also comprising:

delaying the first copy of the input signal in a delay buffer by a configurable delay length to produce a cancellation signal (1120), an

Superimposing the samples contained in the cancellation signal onto the samples contained in the second copy of the input signal (1140); wherein in generating the cancellation signal, the apparatus of any of claims 1 to 9 is further configured to multiply samples from the delay buffer by a coefficient value, wherein the coefficient value effectively inverts the phase of the first copy of the input signal.

18. A computer-readable storage medium, on which a computer program is stored, which, when being executed by a processor, carries out the steps of the method according to any one of claims 13 to 17.